Over the years, more and more users as well as carriers have been switching from circuit switched communications networks to a more efficient and cheaper alternative of packet based communications networks to make telephone calls. For example, technology known as VoIP (Voice over Internet Protocol) is increasing being used to carry voice transmissions.
A key metric used to estimate Quality of Service (QoS) for voice transmissions over a packet network is an end-to-end delay. An important component of the delay is the amount of time it takes for voice data to traverse the network (referred to as “network delay”).
One way to directly measure the network delay between two endpoints of a voice connection would be to synchronize their clocks to a degree of accuracy within a few milliseconds (ms). Voice traffic could then be time-stamped and transmitted. Since both ends of a connection would have common clocks, network delay is a simple calculation of subtracting the timestamp from arrival time. However, the capability to synchronize endpoint clocks to the needed accuracy across a wide area network does not exist in current commercial deployments.
Equipment utilizing the National Institute of Standards Technology (NIST) radio signals (WWV and others) typically have a rated accuracy of 50 ms. Depending on the method used to synchronize time across network devices, Network Time Protocol (NTP) being the most common, accuracy can vary up to 250 ms depending on topology. Global Positioning System (GPS) based clocks have the needed accuracy if they are designed for clock functionality rather than positioning. Additionally current GPS clocks are stationary with stringent open sky requirements and have the same limitations when synchronizing devices across a network as NIST based devices.
In existing commercial deployments of a packet switched (e.g., VoIP) telephone network, there is no direct method to measure the network delay. Instead, the most commonly implemented technique is to estimate the voice network delay by measuring the round trip time of an out-of-band management messages that are used in conjunction with a voice connection with the assumption being that the management messages take the same network path as voice messages. However, accurate estimation using this type of out-of-band messages requires that Real-time Transport Control Protocol (RTCP) be supported. RTCP is an optional protocol used in packet networks to exchange voice quality information.
This method has two major shortcomings. First, in many network configurations, RTCP is not or cannot be supported. Voice connections commonly include both packet and analog circuit switched (i.e., Public Switched Telephone Network) segments. RTCP is strictly a packet network protocol. It is not designed for transmission across analog networks, unlike easily converted voice traffic. Even in a strictly packet topology, RTCP is optional and may not be supported or enabled. The second problem is that there is no guarantee that RTCP messages take the same network path as voice data. Voice data is viewed as real-time, delay intolerant traffic. It is much more likely for a router to assign it a higher priority than a non real-time management message. As a result, estimating the network delay in voice traffic by measuring the roundtrip time of out-of-band management messages is inherently inaccurate.
Therefore, it is desirable to provide a device and method of more accurately estimating the delay of voice transmission over a packet switched communications network.